- 31 May, 2010 2 commits
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Peter Ujfalusi authored
There is no need for the power down wrapper. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Peter Ujfalusi authored
Reset most of the codec registers to their chip reset value. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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- 21 May, 2010 1 commit
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Jorge Eduardo Candelaria authored
When using MCLK is configured for 19.2 Mhz, clock slicer should be enabled and HPPLL should be bypassed in clock path. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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- 20 May, 2010 1 commit
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Jarkko Nikula authored
These pins are for decoupling capacitors for the internal charge pumps in TPA6130A2 and TPA6140A2 and not for connecting external supply. Thanks to Eduardo Valentin <eduardo.valentin@nokia.com> for pointing out the issue with TPA6130A2 and Ilkka Koskinen <ilkka.koskinen@nokia.com> with TPA6140A2. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Reviewed-by: Ilkka Koskinen <ilkka.koskinen@nokia.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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- 19 May, 2010 5 commits
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Jarkko Nikula authored
Codec output pin should be defined with SND_SOC_DAPM_OUTPUT as otherwise external widgets doesn't alter the output state. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Liam Girdwood authored
Fix build warning about unused ops and add ops to the sdp4430 DAI link. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Jorge Eduardo Candelaria authored
Add control to enable earphone driver in TWL6040 codec. This driver is connected to HSDAC Left. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Jorge Eduardo Candelaria authored
Enable earphone speaker in sdp4430 machine driver. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Misael Lopez Cruz authored
Add ASoC support for TI SDP4430. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com> Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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- 17 May, 2010 2 commits
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Peter Ujfalusi authored
Avoid calling the dac33_hard_power when the codec was already in BIAS_OFF state. This could happen in device suspend and module removal time. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Felipe Balbi authored
Since the cases when the same power state would be set again handled gracefully, we do not need to use dev_warn. Signed-off-by: Felipe Balbi <felipe.balbi@nokia.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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- 14 May, 2010 3 commits
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Jorge Eduardo Candelaria authored
McBSP module in OMAP4 needs to be able to set its tx/rx threshold and enable the transmitter/receiver when starting an audio stream. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Tony Lindgren <tony@atomide.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Jorge Eduardo Candelaria authored
In OMAP4, there is only one irq line for TX and RX paths. Use the correct irq line to avoid errors at runtime. Also, request irq line only once (instead of requesting for TX and RX). Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Tony Lindgren <tony@atomide.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Sergey Lapin authored
This patchs should allow to use 32-bit samples on e.g. TLV320AIC3x codec, or others. Signed-off-by: Sergey Lapin <slapin@ossfans.org> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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- 12 May, 2010 1 commit
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Peter Ujfalusi authored
The codec has support for swapping the left and right channels in the digimic interface. New kcontrol to handle this bit. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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- 11 May, 2010 5 commits
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Mark Brown authored
If the FLL is not configured attempting to resume it will produce a warning message so skip the resume. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Mark Brown authored
Disable the output stage prior to the delay stage rather than the other way around. Fixes merge issue with previous headphone output path corrections. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Mark Brown authored
Log the values we're getting back from the DC servo and the values we write to it. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Mark Brown authored
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Peter Ujfalusi authored
If the register for the volume needs invert, than the inversion need to be done from the chip maximum, and not from the platform dependent limit. Introduce soc_mixer_control.platform_max value, which initially equals to chip maximum. The snd_soc_limit_volume function only modify the platform_max, all volsw_info call returns this as well. The .max value holds the chip default (maximum), and it is used for the inversion, if it is needed. Additional check in the volsw_info call has been added to check the validity of the platform_max in case, when custom macros used by codec drivers are not initializing it correctly. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 10 May, 2010 8 commits
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Mark Brown authored
Make dev_() prints much prettier. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Mark Brown authored
This allows more flexible integration with subsystem features. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Mark Brown authored
As well as allowing DAPM pins to be marked as ignoring suspend allow DAI links to be similarly marked. This is primarily intended for digital links between CODECs and non-CPU devices such as basebands in mobile phones and will suppress all suspend calls for the DAI link. It is likely that this will need to be revisited if used with devices which are part of the SoC CPU. Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
Some devices can usefully run audio while the Linux system is suspended. One of the most common examples is smartphone systems, which are normally designed to allow audio to be run between the baseband and the CODEC without passing through the CPU and so can suspend the CPU when on a voice call for additional power savings. Support such systems by providing an API snd_soc_dapm_ignore_suspend(). This can be used to mark DAPM endpoints as not being sensitive to system suspend. When the system is being suspended paths between endpoints which are marked as ignoring suspend will be kept active. Both source and sink must be marked, and there must already be an active path between the two endpoints prior to suspend. When paths are active over suspend the bias management will hold the device bias in the ON state. This is used to avoid suspending the CODEC while it is still in use. Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
Instead of using stream events to handle power down during suspend integrate the handling with the normal widget path checking by replacing all cases where we report a connected endpoint in a path with a function snd_soc_dapm_suspend_check() which looks at the ALSA power state for the card and reports false if we are in a D3 state. Since the core moves us into D3 prior to initating the suspend all power checks during suspend will cause the widgets to be powered down. In order to ensure that widgets are powered up on resume set the card to D2 at the start of resume handling (ALSA API calls require D0 so we are still protected against userspace access). Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
We now manage suspend within the main power analysis rather than by flipping the state of widgets. Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
The core will ensure that the device is in either STANDBY or OFF bias before suspending, restoring the bias in the driver is unneeded. Some drivers doing slightly more roundabout things have been left alone for now. Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 07 May, 2010 7 commits
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Jassi Brar authored
Switch the MACHINE driver to use IISv4 CPU dai. Remove BROKEN dependency now that we have proper CPU driver available. Also, disable build for SMDK6400, since the S3C6400 doesn't have IISv4 controller. Signed-off-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Jassi Brar authored
Add the CPU driver for the IISv4 block found on S3C6410. For now, the driver is almost a copy of s3c64xx-i2s.c but it should diverge as more IISv4 specific stuff is added. Signed-off-by: Jassi Brar <jassi.brar@samsung.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Peter Ujfalusi authored
Since the functions arre only used for volume register, change their name, and also fix them to properly handle the cases, when via soc core the volume is limited. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Peter Ujfalusi authored
This reverts commit 6f399115. Since core has now support for limiting the volume on controls this patch is not needed. Furthermore, this patch actually prevents the core to set new volume on the TPA. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Peter Ujfalusi authored
Add support for the core to limit the maximum volume on an existing control. The function will modify the soc_mixer_control.max value of the given control. The new value must be lower than the original one (chip maximum) If there is a need for limiting a gain on a given control, than machine drivers can do the following in their snd_soc_dai_link.init function: snd_soc_limit_volume(codec, "TPA6140A2 Headphone Playback Volume", 21); This will modify the original 31 (chip maximum) to 21, so user space will not be able to set the gain higher than this. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
Merge branch 'topic/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 into for-2.6.35
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Jassi Brar authored
The S3C DMA API doesn't make use of hw_addr.to/from and also the FIFO addresses are provided from the I2S drivers. So these fields are redundant. This patch removes the hw_addr.to/from fields for I2S and the inclusion of header, paving way for the header to be moved closer to the I2S controller drivers. Signed-off-by: Jassi Brar <jassi.brar@samsung.com> Acked-by: Ben Dooks <ben-linux@fluff.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 06 May, 2010 5 commits
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Takashi Iwai authored
Merge branch 'for-2.6.35' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc
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Peter Ujfalusi authored
Do not change the codec defaults for the following registers: 0x40, 0x41: Line output gains, do not use amplification 0x42: LOM/LOP Voltage hold, and selection 0x44: LOM inversion control It has been found, that the values configured to these registers can cause amplification, which can make the output of DAC33 distorted. The codec reset values are considered safe in all environmnts. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Peter Ujfalusi authored
Add support for platform dependent gain limiting on the tpa6130a2 (and tpa6140a2) Headset amplifier. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Jarkko Nikula authored
Handle the reset GPIO within the codec driver in order to follow the startup protocol for the tlv320aic3x codecs. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Jarkko Nikula authored
This patch adds support for integrated stereo speakers and digital microphone found on Nokia RX-51 hardware. This is a cut down version based on Maemo kernel sources and earlier patchset by Eduardo Valentin et al. http://mailman.alsa-project.org/pipermail/alsa-devel/2009-October/022033.htmlSigned-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Eduardo Valentin <eduardo.valentin@nokia.com> Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Eduardo Valentin <eduardo.valentin@nokia.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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